Grandstream Networks GXP21XX Refer-To Use Target, Contact, Transfer on Conference, Hangup

Page 38

 

Refer-To Use Target

 

Default is “No”. If set to “Yes”, then for Attended Transfer, the “Refer-To” header

 

 

 

 

Contact

 

uses the transferred target’s Contact header information.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Transfer on Conference

 

Defines whether or not the call is transferred to the other party if the initiator of the

 

 

 

 

 

Hangup

 

conference hangs up.

 

 

 

 

 

 

Default setting is set to “No”.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Preferred Vocoder

 

GXP21xx supports up to 7 different Vocoder types including G.711(a/µ) (also known

 

 

 

 

 

 

 

as PCMU/PCMA), G.723.1, G.729A/B, G.726-32, iLBC, G.722 (wide-band).

 

 

 

 

Configure Vocoders in a preference list that is included with the same preference

 

 

 

 

order in SDP message. Enter the first Vocoder in this list by choosing the

 

 

 

 

appropriate option in “Choice 1”. Similarly, enter the last Vocoder in this list by

 

 

 

 

choosing the appropriate option in “Choice 8”.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

SRTP Mode

 

Enable SRTP mode based on selection. Default is “No”.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Symmetric RTP

 

Selects whether or not symmetric RTP is supported.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Silence Suppression

 

This controls the silence suppression/VAD feature of the audio codec G.723 and

 

 

 

 

 

 

 

G.729. If set to “Yes”, when silence is detected, a small quantity of VAD packets

 

 

 

 

(instead of audio packets) will be sent during the period of no talking. If set to “No”,

 

 

 

 

this feature is disabled.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Voice Frames per TX

 

This field contains the number of voice frames to be transmitted in a single Ethernet

 

 

 

 

 

 

 

packet (be advised the IS limit is based on the maximum size of Ethernet packet is

 

 

 

 

1500 byte (or 120kbps)).

 

 

 

 

 

 

When setting this value, be aware of the requested packet time (ptime, used in SDP

 

 

 

 

message) is a result of configuring this parameter. This parameter is associated

 

 

 

 

with the first codec in the above codec Preference List or the actual used payload

 

 

 

 

type negotiated between the 2 conversation parties at run time. E.g., if the first

 

 

 

 

codec is configured as G.723 and the “Voice Frames per TX” is set to 2, then the

 

 

 

 

“ptime” value in the SDP message of an INVITE request will be 60ms because each

 

 

 

 

G.723 voice frame contains 30ms of audio. Similarly, if this field is set to 2 and the

 

 

 

 

first codec is G.729 or G.711 or G.726, then the “ptime” value in the SDP message

 

 

 

 

of an INVITE request will be 20ms.

 

 

 

 

 

 

If the configured voice frames per TX exceeds the maximum allowed value, the IP

 

 

 

 

phone will use and save the maximum allowed value for the corresponding first

 

 

 

 

codec choice. The maximum value for PCM is 10 (x10ms) frames; for G.726, it is 20

 

 

 

 

(x10ms) frames; for G.723, it is 32 (x30ms) frames; for G.729/G.728, 64 (x10ms)

 

 

 

 

and 64 (x2.5ms) frames respectively.

 

 

 

 

 

 

Please be careful when editing these parameters. Adjusting these parameters will

 

 

 

 

also change the dynamic jitter buffer. The GXP21xx has a patent dynamic jitter

 

 

 

 

buffer handling algorithm. The jitter buffer range is 20 ~ 200 ms.

 

 

 

 

We recommend using the default settings provided. We do not recommend

 

 

 

 

adjusting these parameters if you are an average user. Incorrect settings will affect

 

 

 

 

the voice quality.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

No Key Entry Timeout

 

Default is 4 seconds. After the timeout, the phone will send out the dialed number.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Grandstream Networks, Inc.

 

GXP21xx User Manual

Page 37 of 41

 

 

 

 

Firmware version: 1.0.1.66

Last Updated: 05/2011

 

Image 38
Contents Grandstream Networks, Inc Table of Figures GUI Interface Examples Welcome Equipment Packaging InstallationConnecting Your Phone EXTConnecting the GXP2120/2110 and the GXP Extension Safety Compliances WarrantyGXP21xx Comparison Guide Product OverviewGXP21xx Product Models LED GXP21xx Key Features in a GlanceGXP21xx Hardware Specifications Audio Features GXP21xx Technical SpecificationsLines Protocol Support Display Feature Keys HumidityFeatures Telephony FeaturesFirmware Upgrades NetworkLCD Buttons Using the GXP21xx SIP Enterprise PhoneGetting Familiar with the LCD LCD Icons GXP21xx Keypad ButtonsMaking Phone Calls Handset, Speakerphone and Headset ModeMultiple SIP Accounts and Lines Completing Calls Making Calls using IP Addresses Speed DialDo Not Disturb Setting up the phone to make Quick IP callsReceiving Calls For exampleCall Transfer Call Waiting/ Call HoldMute Way ConferencingShared Call Appearance SCA Voice Messages Message Waiting IndicatorBusy Lamp Field End ConferenceCustomized LCD Screen & XML Call FeaturesGXP21xx Call Features Key Pad Configuration Menu Configuration GuideConfiguration VIA Keypad SIP SIP Definitions Access the Web Configuration MenuConfiguration VIA WEB Browser Device Configuration Status Device Configuration Settings/Basic SettingsMTZ+6MDT+5 Currency Update Weather UpdateStock Update Disable in-call DtmfDevice Configuration Settings /Advanced Settings ACS URL Download At Boot-upPort Phonebook XML Phonebook XML ServerSend SIP Log Password Distinctive Ring ToneSyslog Level NTP serverDtmf SIP Account Settings DNS Mode Authenticate IDAuthenticate Password Backup IPPublish for Presence NAT TraversalSubscribe for MWI Proxy-RequireDial Plan Delayed Call ForwardEnable Call Features Wait TimeForce Timer Caller Request TimerCallee Request Timer Enable 100relRefer-To Use Target Srtp ModeNo Key Entry Timeout ContactSaving the Configuration Changes Rebooting the Phone RemotelyFirmware Upgrade Through TFTP/HTTP Software Upgrade & CustomizationWeb Configuration Interface Key Pad MenuConfiguration File Download Managing Firmware and Configuration File DownloadInstructions for Local Tftp Upgrade Restore Factory Default Setting Instructions for Restoration
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GXP21XX specifications

Grandstream Networks GXP21XX series is a line of high-performance VoIP (Voice over Internet Protocol) desktop IP phones designed to deliver a versatile communication experience for both small and medium-sized businesses as well as large enterprises. The GXP21XX series encompasses a range of models, including the GXP2130, GXP2140, and GXP2160, each offering distinct features and capabilities to cater to various user needs.

One of the main features of the GXP21XX series is its support for high-definition (HD) voice technology. This ensures crystal-clear audio quality during calls, enhancing the overall communication experience. HD voice is complemented by advanced echo cancellation and noise reduction technologies, making it easy to communicate even in bustling environments.

The GXP21XX series boasts a user-friendly interface supported by a large LCD display, which is designed for easy navigation and menu access. These phones offer programmable keys, allowing users to customize their devices for quick access to frequently used features, including speed dials and call transfers.

In terms of connectivity, the GXP21XX series is equipped with dual Gigabit Ethernet ports, facilitating high-speed connections to both the local network and the internet. This allows for seamless integration into existing infrastructures without compromising call quality or network performance. Additionally, the phones support Power over Ethernet (PoE), enabling straightforward installation without the need for additional power sources.

The phones come pre-loaded with a multitude of voice codecs, including G.722, G.711, and G.729, offering flexibility to adapt to various network conditions. Users can enjoy secure communication through built-in security encryption features, such as TLS and SRTP, ensuring sensitive data remains safeguarded.

Moreover, the GXP21XX series incorporates advanced functionalities, such as support for up to six SIP accounts, call presence, and a full-duplex speakerphone, making them ideal for multi-tasking and collaborative environments. Interoperability with major VoIP platforms enhances compatibility, ensuring that users can easily integrate their devices into diverse communication systems.

In conclusion, the Grandstream GXP21XX series stands out as a robust and feature-rich solution for VoIP communication. With its focus on HD voice quality, user-friendly interface, and advanced connectivity options, it provides a comprehensive tool for enhancing both individual and team productivity. Whether used in an office environment or for remote work, the GXP21XX series delivers reliability and performance tailored to the needs of today’s business communication landscape.